#
# @description Pre-test: proxy run; Register (use SIP registration) device that is configured as a PBX device in sim-ring profile; ensure it's registered using JTAPI,<br />1. Half calls  send a 100  (run on ports 5081-proxy port defined according to sub profile);<br />2. Full and PBX calls receive INVITE  with CSE sessionID in "From" &amp; "PAID"    (Full run on port 5061 - call routed to this port by Route: &lt;sip:[field0]@[local_ip]:5061;lr&gt;; PBX uas run on port 5088 according to sub profile)<br />3. As result all half calls Cancelled  and Full call  sends 100, 200(INVITE), receives ACK, BYE, send 200(BYE)
# @sincerelease 3.1
#
#SIPp uas_SIMRING12_A.xml dat_sr_pbx_csn02.csv 5062
#SIPp uas_SIMRING12_NA.xml dat_sr_pbx_csn02.csv 5081
#SIPp uas_SIMRING12_PBX.xml dat_sr_pbx_csn02.csv 5088
sleep 2
#SIPp uac_SIMRING12.xml dat_sr_pbx_csn02.csv
